A unified approach to tracking performance analysis of the selective partial update adaptive filter algorithms in nonstationary environment

In this paper, a unified approach to mean-square performance analysis of the family of selective partial update (SPU) adaptive filter algorithms in nonstationary environment is presented. Using this analysis, the tracking performance of Max normalized least mean squares (Max-NLMS), N-Max NLMS, the various types of SPU-NLMS algorithms, SPU transform domain LMS (SPU-TD-LMS), the family of SPU affine projection algorithms (SPU-APA), the family of selective regressor APA (SR-APA), the dynamic selection of APA (DS-APA), the family of SPU-SR-APA, the family of SPU-DS-APA, SPU subband adaptive filters (SPU-SAF), and the periodic, sequential, and stochastic partial update LMS, NLMS, and APA as well as classical adaptive filter algorithms can be analyzed with a unified approach. Two theoretical expressions are introduced to study the performance. The analysis is based on energy conservation arguments and does not need to assume a Gaussian or white distribution for the regressors. We demonstrate through simulations that the derived expressions are useful in predicting the performance of this family of adaptive filters in nonstationary environment.     

A diffusion subband adaptive filtering algorithm for distributed estimation using variable step size and new combination method based on the MSD

This paper proposes a novel diffusion subband adaptive filtering algorithm for distributed networks. To achieve a fast convergence rate and small steady-state errors, a variable step size and a new combination method is developed. For the adaptation step, the upper bound of the mean-square deviation (MSD) of the algorithm is derived and the step size is adaptive by minimizing it in order to attain the fastest convergence rate on every iteration. Furthermore, for a combination step realized by a convex combination of the neighbor-node estimates, the proposed algorithm uses the MSD, which contains information on the reliability of the estimates, to determine combination coefficients. Simulation results show that the proposed algorithm outperforms the existing algorithms in terms of the convergence rate and the steady-state errors.     

基于集员滤波的选择权更新子带有源噪声控制

通过子带自适应滤波结构,可以提高宽带噪声降噪效果,归一化子带自适应滤波（NSAF）通过在每个子带上使用相同的全带自适应滤波器,消除了传统子带结构会在输出端产生混叠分量的问题,具有较好的收敛性能和稳态均方误差。但由于在每个子带上采用相同的全带自适应滤波器,计算量要高于传统子带结构,集员滤波（SMF）技术具有数据选择更新的特点,可有效降低计算复杂度。基于NSAF结构,建立了前馈ANC无延迟结构,并基于集员滤波技术,通过选择部分权更新来进一步减少计算量,仿真验证了该算法对宽带噪声具有更优的降噪效果。     

Diffusion LMS algorithms with multi combination for distributed estimation: Formulation and performance analysis

We propose diffusion least-mean-square (LMS) algorithms that use multi-combination step. We allow each node in the network to use information from multi-hop neighbors to approximate a global cost function accurately. By minimizing this cost and dividing multi-hop range summation into 1-hop range combination steps, we derive new diffusion LMS algorithms. The resulting distributed algorithms consist of adaptation and multi-combination step. Multi combination allows each node to use information from non-adjacent nodes at each time instant, thereby reducing steady-state error. We analyzed the output to derive stability conditions and to quantify the transient and steady-state behaviors. Theoretical and experimental results indicate that the proposed algorithms have lower steady-state error compared to the conventional diffusion LMS algorithms. We also propose a new combination rule for the multi-combination step which can further improve the estimation performance of the proposed algorithms.     

Subband neural networks prediction for on-line audio signal recovery

In this paper, a subbands multirate architecture is presented for audio signal recovery. Audio signal recovery is a common problem in digital music signal restoration field, because of corrupted samples that must be replaced. The subband approach allows for the reconstruction of a long audio data sequence from forward-backward predicted samples. In order to improve prediction performances, neural networks with spline flexible activation function are used as narrow subband nonlinear forward-backward predictors. Previous neural-networks approaches involved a long training process. Due to the small networks needed for each subband and to the spline adaptive activation functions that speed-up the convergence time and improve the generalization performances, the proposed signal recovery scheme works in online (or in continuous learning) mode as a simple nonlinear adaptive filter. Experimental results show the mean square reconstruction error and maximum error obtained with increasing gap length, from 200 to 5000 samples for different musical genres. A subjective performances analysis is also reported. The method gives good results for the reconstruction of over 100 ms of audio signal with low audible effects in overall quality and outperforms the previous approaches.     

A New Delayless Subband Adaptive Filtering Algorithm for Active Noise Control Systems

Subband adaptive filtering (SAF) techniques play a prominent role in designing active noise control (ANC) systems. They reduce the computational complexity of ANC algorithms, particularly, when the acoustic noise is a broadband signal and the system models have long impulse responses. In the commonly used uniform-discrete Fourier transform (DFT) -modulated (UDFTM) filter banks, increasing the number of subbands decreases the computational burden but can introduce excessive distortion, degrading performance of the ANC system. In this paper, we propose a new UDFTM-based adaptive subband filtering method that alleviates the degrading effects of the delay and side-lobe distortion introduced by the prototype filter on the system performance. The delay in filter bank is reduced by prototype filter design and the side-lobe distortion is compensated for by oversampling and appropriate stacking of subband weights. Experimental results show the improvement of performance and computational complexity of the proposed method in comparison to two commonly used subband and block adaptive filtering algorithms.      

On adaptive inverse dynamics for free-floating space manipulators

This paper is devoted to the investigation of adaptive inverse dynamics for free-floating space manipulators (FFSMs) suffering from parameter uncertainties/variations. To overcome the nonlinear parametric problem of the dynamics of FFSMs, we introduce a new regressor matrix called the generalized dynamic regressor. Based on this regressor, and with Lyapunov stability analysis tools, we obtain a new parameter adaptation law and show that the closed-loop system is stable, and that the joint tracking errors converge asymptotically to zero. Simulation results are provided to illustrate the performance of the proposed adaptive algorithm. Furthermore, we conduct a comparative study between adaptive inverse dynamics, prediction error based adaptation, and passivity based adaptation.     

A hierarchical alternative updated adaptive Volterra filter with pipelined architecture

The pipelined adaptive Volterra filters (PAVFs) with a two-layer structure constitute a class of good low-complexity filters. They can efficiently reduce the computational complexity of the conventional adaptive Volterra filter. Their major drawbacks are low convergence rate and high steady-state error caused by the coupling effect between the two layers. In order to remove the coupling effect and improve the performance of PAVFs, we present a novel hierarchical pipelined adaptive Volterra filter (HPAVF)-based alternative update mechanism. The HPAVFs with hierarchical decoupled normalized least mean square (HDNLMS) algorithms are derived to adaptively update weights of its nonlinear and linear subsections. The computational complexity of HPAVF is also analyzed. Simulations of nonlinear system adaptive identification, nonlinear channel equalization, and speech prediction show that the proposed HPAVF with different independent weight vectors in nonlinear subsection has superior performance to conventional Volterra filters, diagonally truncated Volterra filters, and PAVFs in terms of initial convergence, steady-state error, and computational complexity.     

Forward stage‐wise ensemble regression algorithm to improve base regressors prediction ability: an empirical study

Analysis of scientific data requires accurate regressor algorithms to decrease prediction errors. Lots of machine learning algorithms, that is, neural networks, rule‐based algorithms, regression trees and some kinds of lazy learners, are used to realize this need. In recent years, different ensemble regression strategies were improved to obtain enhanced predictors with lower forecasting errors. Ensemble algorithms combine good models that make errors in different parts of analyzed data. There are mainly two approaches in ensemble regression algorithm generation; boosting and bagging. The aim of this article is to evaluate a boosting‐based ensemble approach, forward stage‐wise additive modelling (FSAM), to improve some widely used base regressors’ prediction ability. We used 10 regression algorithms in four different types to make predictions on 10 diverse data from different scientific areas and we compared the experimental results in terms of correlation coefficient, mean absolute error, and root mean squared error metrics. Furthermore, we made use of scatter plots to demonstrate the effect of ensemble modelling on the prediction accuracies of evaluated algorithms. We empirically obtained that in general FSAM enhances the accuracies of base regressors or it at least maintains the base regressor performance.     

用于稀疏系统辨识的变步长加权零吸引最小平均p范数算法

在α稳定分布噪声背景下,为了提高稀疏系统自适应辨识算法的稳态性能,提出了基于无噪先验误差功率函数的变步长加权零吸引最小平均p范数基本算法(BVSS-RZA-LMP)和变步长加权零吸引最小平均p范数改进算法(IVSS-RZA-LMP).两种算法分别根据无噪先验误差功率和加权的无噪先验误差功率计算新的步长;步长随无噪先验误差功率的减小而逐渐减小.当算法达到稳态时, IVSS-RZA-LMP算法不再调整权矢量,改进了BVSSRZA-LMP算法稳态性能.α稳定分布噪声背景下的系统辨识仿真结果表明,当系统较稀疏时, IVSS-RZA-LMP算法能够在较快收敛的情况下获得非常小的稳态误差.     

扩散式变阶数最大相关熵准则算法

固定阶数的分布式自适应滤波算法只有在待估计向量的阶数已知且恒定的情况下才能达到相应的估计精度,在阶数未知或时变的情况下算法的收敛性能会受到影响,变阶数的分布式自适应滤波算法是解决上述问题的有效途径。但是目前大多数分布式变阶数自适应滤波算法以最小均方误差(Mean square Error,MSE)准则作为滤波器阶数的代价函数,在脉冲噪声环境下算法的收敛过程会受到较大影响。最大相关熵准则具有对脉冲噪声的强鲁棒性,且计算复杂度低。为提高分布式变阶数自适应滤波算法在脉冲噪声环境下的估计精度,利用最大相关熵准则作为滤波器阶数迭代的代价函数,并将得到的结果代入固定阶数的扩散式最大相关熵准则算法,提出了一种扩散式变阶数最大相关熵准则(Diffusion Variable Tap-length Maximum Correntropy Criterion,DVTMCC)算法。通过与邻域的节点进行通信,所提算法以扩散的方式实现了整个网络的信息融合,具有估计精度高、计算量小等优点。仿真实验对比了在脉冲噪声下DVTMCC算法和其他分布式变阶数自适应滤波算法、固定阶数的扩散式最大相关熵准则算法的收敛性能。仿真结果表明,在脉冲噪声环境下DVTMCC算法能够同时估计未知向量的阶数和权值,性能优于参与对比的算法。     

A comparative analysis of evolutionary and memetic algorithms for community detection from signed social networks

To detect communities in signed networks consisting of both positive and negative links, two new evolutionary algorithms (EAs) and two new memetic algorithms (MAs) are proposed and compared. Furthermore, two measures, namely the improved modularity Q and the improved modularity density D-value, are used as the objective functions. The improved measures not only preserve all properties of the original ones, but also have the ability of dealing with negative links. Moreover, D-value can also control the partition to different resolutions. To fully investigate the performance of these four algorithms and the two objective functions, benchmark social networks and various large-scale randomly generated signed networks are used in the experiments. The experimental results not only show the capability and high efficiency of the four algorithms in successfully detecting communities from signed networks, but also indicate that the two MAs outperform the two EAs in terms of the solution quality and the computational cost. Moreover, by tuning the parameter in D-value, the four algorithms have the multi-resolution ability.     

未知控制方向的迟滞非线性系统预设自适应控制

针对一类含有迟滞特性的未知控制方向严反馈非线性系统,设计了基于误差变换的反步自适应控制器.首先提出动态迟滞算子来扩展输入空间建立神经网络迟滞模型.然后利用径向基函数(RBF)神经网络逼近未知函数,并引入Nussbaum型函数来解决系统未知控制方向问题.最后采用误差变换将误差限定在预设的范围内,并利用反步法设计自适应控制器.该控制方案不仅能够保证跟踪精度,还可以提高系统暂态和稳态性能.仿真结果表明了控制方案的可行性.     

Mean-square error and stability analysis of a subband structure for the rapid identification of sparse impulse responses

In a context of supervised adaptive filtering, the sparsity of the impulse response to be identified can be employed to accelerate the convergence rate of the algorithm. This idea was first explored by the so-called proportionate NLMS (PNLMS) algorithm, where the adaptation step-sizes are made larger for the coefficients with larger magnitudes. Whereas fast initial adaptation convergence rate is obtained with the PNLMS algorithm for white-noise input, slow convergence is observed for colored input signals. The combination of the PNLMS approach and a subband structure results in an algorithm with better convergence rate for sparse systems and colored input signals. In this paper, the steady-state mean-square error (MSE) and the maximum value of the step-size β that allows convergence of the subband PNLMS-type algorithm are analyzed. Theoretical results are confirmed by simulations.     

Improved subband-forward algorithm for acoustic noise reduction and speech quality enhancement

This paper addresses the problem of speech enhancement and acoustic noise reduction by adaptive filtering algorithms. Recently, we have proposed a new Forward blind source separation algorithm that enhances very noisy speech signals with a subband approach. In this paper, we propose a new variable subband step-sizes algorithm that allows improving the previous algorithm behaviour when the number of subband is selected high. This new proposed algorithm is based on recursive formulas to compute the new variable step-sizes of the cross-coupling filters by using the decorrelation criterion between the estimated sub-signals at each subband output. This new algorithm has shown an important improvement in the steady state and the mean square error values. Along this paper, we present the obtained simulation results by the proposed algorithm that confirm its superiority in comparison with its original version that employs fixed step-sizes of the cross-coupling adaptive filters and with another fullband algorithm.     

An improved 3D wavelet-based scalable video coding codec for MC-EZBC

With the rapid growth of modern multimedia applications, 3D wavelet-based scalable video coding (SVC) codec has received considerable attention lately because of its high coding performance and flexibility in bitstream scalability. It combines the motion-compensated temporal filtering (MCTF) together with the spatial decomposition to produce an embedded bitstream offering various levels of video quality over the heterogeneous networks. However, in the existing 3D wavelet-based SVC schemes, where the block types for block matching algorithms are limited, weighting matrices for block-wise motion compensation are fixed, and variations in activities of temporal subbands are not considered in the selection of the Lagrange multiplier for mode decision. In this paper, our major contribution is to provide some recent extensions to the well-known scalable subband/wavelet video codec Motion-Compensated Embedded Zero Block Coding (MC-EZBC) using three novel and content adaptive algorithms. Firstly, the enhanced hierarchical variable size block matching (Enhanced HVSBM) algorithm is proposed for the variable block size motion estimation. Then, the rate-distortion optimization (RDO) based adaptive Lagrange multiplier selection model for mode decision is presented. Finally, we introduce the adaptive weighting matrices design for overlapped block motion compensation (OBMC). Experimental results show that all the three proposed algorithms significantly improve the overall coding performance of MC-EZBC. Comparisons with other popular wavelet-based SVC codecs demonstrate the effectiveness of our improved codec in terms of both video quality assessment and computational complexity.     

基于最大后验估计的自适应盲均衡算法

为了减小最大后验概率(MAP)盲均衡算法中稳态误差和收敛速度,提出了一种改进的基于最大后验估计的盲均衡算法.运用不等概思想改进自适应σ的MAP盲均衡算法,得到了基于概率和自适应σ的MAP盲均衡算法.定量分析和仿真结果表明改进算法和MAP的盲均衡算法相比,稳态调整量的理论值更小,稳态剩余根均方误差值小于MAP盲均衡算法,收敛速度快于MAP盲均衡算法.     

Diffusion maximum correntropy criterion algorithms for robust distributed estimation

Robust diffusion adaptive estimation algorithms based on the maximum correntropy criterion (MCC), including adapt then combine MCC and combine then adapt MCC, are developed to deal with the distributed estimation over network in impulsive (long-tailed) noise environments. The cost functions used in distributed estimation are in general based on the mean square error (MSE) criterion, which is desirable when the measurement noise is Gaussian. In non-Gaussian situations, especially for the impulsive-noise case, MCC based methods may achieve much better performance than the MSE methods as they take into account higher order statistics of error distribution. The proposed methods can also outperform the robust diffusion least mean p-power (DLMP) and diffusion minimum error entropy (DMEE) algorithms. The mean and mean square convergence analysis of the new algorithms are also carried out.     

分布式小波视频编码自适应嵌套量化模型

提出一种分布式小波视频编码自适应嵌套量化模型,对Wyner-Ziv帧做小波变换,利用Slepian—Wolf编码器对低频子带系数进行量化,采用SPIHT编码器对高频子带系数进行编码,通过分析Wyner—Ziv帧低频子带系数与参考帧低频子带系数的残差自适应度,判定Wyner-Ziv帧低频子带系数的量化步长,从而使运动剧烈的视频序列有较大的量化步长,而运动平缓序列的量化步长较小。实验结果表明,采用该模型能获得较好的编码效果。     

Least mean square algorithms with Markov regime-switching limit

We analyze the tracking performance of the least mean square (LMS) algorithm for adaptively estimating a time varying parameter that evolves according to a finite state Markov chain. We assume the Markov chain jumps infrequently between the finite states at the same rate of change as the LMS algorithm. We derive mean square estimation error bounds for the tracking error of the LMS algorithm using perturbed Lyapunov function methods. Then combining results in two-time-scale Markov chains with weak convergence methods for stochastic approximation, we derive the limit dynamics satisfied by continuous-time interpolation of the estimates. Unlike most previous analyzes of stochastic approximation algorithms, the limit we obtain is a system of ordinary differential equations with regime switching controlled by a continuous-time Markov chain. Next, to analyze the rate of convergence, we take a continuous-time interpolation of a scaled sequence of the error sequence and derive its diffusion limit. Somewhat remarkably, for correlated regression vectors we obtain a jump Markov diffusion. Finally, two novel examples of the analysis are given for state estimation of hidden Markov models (HMMs) and adaptive interference suppression in wireless code division multiple access (CDMA) networks.